The public switched telephone network (PSTN) has traditionally provided telephony communications to the masses. The PSTN is a circuit-switched network where each call is essentially allocated a dedicated circuit through which audio signals are carried. These audio signals may include voice information as well as in-band signaling information. The in-band signaling information is tones or tone sequences for providing audible alerts to one of the parties to the call as well as for conveying control information among communication terminals and call processing entities in the PSTN. For example, the audible alerts may include the traditional busy, fast busy, and ringing alerts provided to a caller when the call is being initiated. The control information may include dual tone multi-frequency (DTMF) tones corresponding to keys on the telephone terminal's keypad. The DTMF tones may correspond to dialed digits used to initiate a call or a selection made by a caller during automated call processing. The events requiring delivery of in-band signaling information are generally referred to as named signal events.
Given the increased capacity and reliability of packet networks, such as the Internet, telephony communications can now be supported over the packet networks. Packet-based telephony communications are often referred to as voice-over-packet (VoP) communications or, when supported by the Internet Protocol (IP), voice over IP (VoIP) communications. For VoP telephony, audio signals are encoded and placed into packets, which are delivered over the packet networks.
In an effort to maintain a consistent user experience among PSTN and VoP telephony, named signal events are used VoP telephony. However, in VoP telephony the tones associated with the named signal events are generally not encoded with the voice information. Instead, a tone description corresponding to the tone is placed in a packet, which is delivered over the packet network. The tone description may define the tone or tone sequence as well as length information describing how long the each tone should be provided when presented to the intended party or device. Further information regarding the handling of named signal events for VoP telephony is provided in Internet Engineering Task Force (IETF) Request For Comment (RFC) 2833, which is incorporated herein by reference.
Notably, VoP packets for telephony communications are streamed from one communication terminal to another over the packet network in real time. Since there is no time for retransmission of lost or damaged packets and the nature of voice allows for a significant number of lost or damaged packets without undue degradation of the voice signal, error checking is not provided for received voice packets. When packet networks provide wireless access for wireless communication terminals, the likelihood of damaged packets being received increases dramatically over wired packet networks. The impact of damaged packets on the voice signal is tolerable; however, the impact on packets carrying named signal event information is more problematic. Errors may result in generating the wrong tones, generating tones for the wrong lengths of time, and the like. Since error checking is often disabled for VoIP telephony in a wireless environment, there is no way to detect the errors in the packets carrying named signal event information. Accordingly, there is need for a way to efficiently and effectively detect and process errant packets carrying named signal event information in association with a VoIP telephony call.